Index: media/audio/win/audio_low_latency_output_win.h |
diff --git a/media/audio/win/audio_low_latency_output_win.h b/media/audio/win/audio_low_latency_output_win.h |
index fb9aa3dd074ad659f44b3df2a3cc12622ec7015d..d32a286f391aac04aa173655b2a9c0700bf931db 100644 |
--- a/media/audio/win/audio_low_latency_output_win.h |
+++ b/media/audio/win/audio_low_latency_output_win.h |
@@ -21,17 +21,10 @@ |
// render thread and at the same time stops audio streaming. |
// - The same thread that called stop will call Close() where we cleanup |
// and notify the audio manager, which likely will destroy this object. |
-// - Initial tests on Windows 7 shows that this implementation results in a |
-// latency of approximately 35 ms if the selected packet size is less than |
-// or equal to 20 ms. Using a packet size of 10 ms does not result in a |
-// lower latency but only affects the size of the data buffer in each |
-// OnMoreData() callback. |
// - A total typical delay of 35 ms contains three parts: |
// o Audio endpoint device period (~10 ms). |
// o Stream latency between the buffer and endpoint device (~5 ms). |
// o Endpoint buffer (~20 ms to ensure glitch-free rendering). |
-// - Note that, if the user selects a packet size of e.g. 100 ms, the total |
-// delay will be approximately 115 ms (10 + 5 + 100). |
// |
// Implementation notes: |
// |
@@ -39,18 +32,11 @@ |
// - This implementation is single-threaded, hence: |
// o Construction and destruction must take place from the same thread. |
// o All APIs must be called from the creating thread as well. |
-// - It is recommended to first acquire the native sample rate of the default |
-// input device and then use the same rate when creating this object. Use |
-// WASAPIAudioOutputStream::HardwareSampleRate() to retrieve the sample rate. |
+// - It is required to first acquire the native audio parameters of the default |
+// output device and then use the same rate when creating this object. Use |
+// e.g. WASAPIAudioOutputStream::HardwareSampleRate() to retrieve the sample |
+// rate. Open() will fail unless "perfect" audio parameters are utilized. |
// - Calling Close() also leads to self destruction. |
-// - Stream switching is not supported if the user shifts the audio device |
-// after Open() is called but before Start() has been called. |
-// - Stream switching can fail if streaming starts on one device with a |
-// supported format (X) and the new default device - to which we would like |
-// to switch - uses another format (Y), which is not supported given the |
-// configured audio parameters. |
-// - The audio device must be opened with the same number of channels as it |
-// supports natively (see HardwareChannelCount()) otherwise Open() will fail. |
// - Support for 8-bit audio has not yet been verified and tested. |
// |
// Core Audio API details: |
@@ -183,22 +169,13 @@ class MEDIA_EXPORT WASAPIAudioOutputStream : |
// Issues the OnError() callback to the |sink_|. |
void HandleError(HRESULT err); |
- // The Open() method is divided into these sub methods. |
- HRESULT SetRenderDevice(); |
- HRESULT ActivateRenderDevice(); |
- bool DesiredFormatIsSupported(); |
- HRESULT InitializeAudioEngine(); |
- |
- // Called when the device will be opened in shared mode and use the |
- // internal audio engine's mix format. |
- HRESULT SharedModeInitialization(); |
- |
// Called when the device will be opened in exclusive mode and use the |
// application specified format. |
- HRESULT ExclusiveModeInitialization(); |
- |
- // Converts unique endpoint ID to user-friendly device name. |
- std::string GetDeviceName(LPCWSTR device_id) const; |
+ // TODO(henrika): rewrite and move to CoreAudioUtil when removing flag |
+ // for exclusive audio mode. |
+ HRESULT ExclusiveModeInitialization(IAudioClient* client, |
+ HANDLE event_handle, |
+ size_t* endpoint_buffer_size); |
// Contains the thread ID of the creating thread. |
base::PlatformThreadId creating_thread_id_; |
@@ -215,25 +192,17 @@ class MEDIA_EXPORT WASAPIAudioOutputStream : |
// Use this for multiple channel and hi-resolution PCM data. |
WAVEFORMATPCMEX format_; |
- // Copy of the audio format which we know the audio engine supports. |
- // It is recommended to ensure that the sample rate in |format_| is identical |
- // to the sample rate in |audio_engine_mix_format_|. |
- base::win::ScopedCoMem<WAVEFORMATPCMEX> audio_engine_mix_format_; |
- |
+ // Set to true when stream is successfully opened. |
bool opened_; |
- // Set to true as soon as a new default device is detected, and cleared when |
- // the streaming has switched from using the old device to the new device. |
- // All additional device detections during an active state are ignored to |
- // ensure that the ongoing switch can finalize without disruptions. |
- bool restart_rendering_mode_; |
+ // We check if the input audio parameters are identical (bit depth is |
+ // excluded) to the preferred (native) audio parameters during construction. |
+ // Open() will fail if |audio_parmeters_are_valid_| is false. |
+ bool audio_parmeters_are_valid_; |
// Volume level from 0 to 1. |
float volume_; |
- // Size in bytes of each audio frame (4 bytes for 16-bit stereo PCM). |
- size_t frame_size_; |
- |
// Size in audio frames of each audio packet where an audio packet |
// is defined as the block of data which the source is expected to deliver |
// in each OnMoreData() callback. |
@@ -256,11 +225,6 @@ class MEDIA_EXPORT WASAPIAudioOutputStream : |
// where AUDCLNT_SHAREMODE_SHARED is the default. |
AUDCLNT_SHAREMODE share_mode_; |
- // The channel count set by the client in |params| which is provided to the |
- // constructor. The client must feed the AudioSourceCallback::OnMoreData() |
- // callback with PCM-data that contains this number of channels. |
- int client_channel_count_; |
- |
// Counts the number of audio frames written to the endpoint buffer. |
UINT64 num_written_frames_; |
@@ -270,9 +234,6 @@ class MEDIA_EXPORT WASAPIAudioOutputStream : |
// An IMMDeviceEnumerator interface which represents a device enumerator. |
base::win::ScopedComPtr<IMMDeviceEnumerator> device_enumerator_; |
- // An IMMDevice interface which represents an audio endpoint device. |
- base::win::ScopedComPtr<IMMDevice> endpoint_device_; |
- |
// An IAudioClient interface which enables a client to create and initialize |
// an audio stream between an audio application and the audio engine. |
base::win::ScopedComPtr<IAudioClient> audio_client_; |